I like the idea that we could maintain a local registrar that accurately
reflects the remote registration. I would go so far as to say that it would
be useful to be able to optimize opensips as an ALG. Some people could use a
proxy as an edge device on a LAN. You could have several phones with the same
user name register with a service provider. Some service providers limit the
number of contacts per AOR. This would have the side effect of keeping the
signaling associated with forking on the LAN. By detecting local calls, media
could also be kept on the LAN. Such an edge device might have a dynamic IP
address. It would be helpful if opensips could conveniently be setup to fix
the contacts. Using the received address does not work when the phones are on
the same LAN as the proxy.
Opensips has a bias toward service providers. This is quite understandable.
However I feel with a bit of tweaking it could serve as an ALG for the home
owner or a small business.
On Wednesday 13 May 2009, Bogdan-Andrei Iancu wrote:
> Hi Jeff,
>
> theoretically yes, because you have all the needed information
> (hmm...maybe except the NAT status from request, but you can store it
> via transaction)....Practically, the save() function does expect to
> receive a request only, so it must be changed to work with a reply also.
>
> Regards,
> Bogdan
>
> Jeff Pyle wrote:
> > I've thought a lot about this as well, although I haven't taken it nearly
> > as far as John has.
> >
> > A thought: is it possible to do a save() in the reply route, only upon a
> > 200 OK from the end registrar?
> >
> >
> > - Jeff
> >
> > On 5/12/09 5:09 AM, "Bogdan-Andrei Iancu" <bog...@voice-system.ro> wrote:
> >> Hi John,
> >>
> >> This mid-registrar approach may work but it is not 100% correct as
> >> OpenSIPS (as mid-registrar) does not obey the actions of the final
> >> registrar (Asterisk). Ex:
> >> - Asterisk may forbid the registration and you already saved the
> >> registration on OpenSIPS
> >> - Asterisk may change the Expire time while to saved the
> >> registration with the expire sent by client.
> >>
> >> Anyhow, ignoring this aspects, lets go further :) :
> >>
> >> 1) is the registration scenario working ok? if not what is the exact
> >> problem (some trace will help).
> >>
> >> I will wait for you answer before moving further with the calling stuff.
> >>
> >> Regards,
> >> Bogdan
> >>
> >> John Morris wrote:
> >>> After several days of playing with OpenSIPS 1.5.0 and RTPProxy 1.2.0, I
> >>> have a partially working SIP+RTP ALG configuration, and have gotten
> >>> stuck. I could use some general advice from the list.
> >>>
> >>> The company has an Asterisk/FreePBX server on an internal network, and
> >>> the CEO wants to use a SIP phone from outside. Because the sip alg
> >>> iptables module isn't working, and in preparation for another project,
> >>> I started investigating OpenSIPS for use as a border proxy to connect
> >>> phones across NAT (and, the next project, to route a SIP trunk over a
> >>> VPN from the network of a DSL+phone company that intermittently blocks
> >>> SIP traffic in hopes of plugging revenue leaks).
> >>>
> >>> The network looks like this:
> >>>
> >>> SIP UA <-> home NAT gateway <-> Internet <-> OpenSIPS server/NAT router
> >>> <-> Asterisk
> >>>
> >>> The standard opensips.cfg file doesn't work as is. The SIP phone needs
> >>> to register to the Asterisk server directly. In addition, it seems
> >>> there is extra logic needed to support multiple network interfaces
> >>> (mhomed=1 only partially solves the problem).
> >>>
> >>> The way I've gone with this in testing is to relay REGISTERs to
> >>> Asterisk, but after a save("location","0x02") to enable a
> >>> lookup("location") on messages originating from the PBX. The phone is
> >>> configured with an outbound proxy, and all packets to the proxy
> >>> matching "uri==myself" are thrown away. This worked great on the
> >>> single-interfaced, internal test installation. Now that there are
> >>> multiple interfaces involved, things are breaking again; ACKs and BYEs
> >>> are sent out the wrong interface, and RTPProxy is behaving strangely in
> >>> bridged mode.
> >>>
> >>> There seem to be no good configuration examples for either multi-homed
> >>> proxies or for proxies that relay REGISTERs. This makes me think that
> >>> I'm going about this the wrong way.
> >>>
> >>> Also, I have looked at other software, like siproxd, opensbc and uh,
> >>> that other b2bua that functions as an SBC, but none of those seem to
> >>> allow this REGISTER pass-through function.
> >>>
> >>> What is the best approach for this scenario? The above approach of
> >>> relaying REGISTERs to Asterisk? Is there maybe another approach where
> >>> phones register to OpenSIPS directly, and OpenSIPS in turn somehow
> >>> sends another REGISTER to Asterisk? Or am I missing the idea
> >>> completely?
> >>>
> >>> I'd appreciate general pointers about how to proceed. I've been
> >>> putting some Asterisk and FreePBX tutorials and CentOS RPMs on
> >>> http://www.zultron.com, mostly aimed at small office-like environments.
> >>> Looking through various lists, this seems a highly sought-after
> >>> configuration. If I succeed, I'll document it in hopes of filling the
> >>> gap in this sort of example.
> >>>
> >>> Thanks
> >>>
> >>> John
> >>>
> >>>
> >>> _______________________________________________
> >>> Users mailing list
> >>> Users@lists.opensips.org
> >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>
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>
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